draft-ietf-rtcweb-rtp-usage
IETF DataTracker: draft-ietf-rtcweb-rtp-usage
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Media Transport and Use of RTP in WebRTC, Internet Engineering Task Force, RFC 8834, January 2021. DOI:10.17487/RFC8834
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, March 2016, Work in progress (draft-ietf-rtcweb-rtp-usage-26.txt).
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This version makes two minor changes. Firstly, as discussed at IETF 94, in Section 4.5, clarify that support for sending RTP and RTCP using two separate transport layer flows is OPTIONAL. Secondly, change RTP Packet Stream to RTP Stream throughout, to align terminology with RFC 7656.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, June 2015, Work in progress (draft-ietf-rtcweb-rtp-usage-25.txt).
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Numerous minor fixes and clarifications to address the security directorate review and area director review comments from Spencer Dawkins and Ben Campbell.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, May 2015, Work in progress (draft-ietf-rtcweb-rtp-usage-24.txt).
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Minor clarifications to address IETF Area Director review comments.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, March 2015, Work in progress (draft-ietf-rtcweb-rtp-usage-23.txt).
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Section 7.2 is updated to mandate RTCP implementation by all endpoints that wish to interwork with WebRTC devices, since it is needed for congestion control purposes.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, February 2015, Work in progress (draft-ietf-rtcweb-rtp-usage-22.txt).
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This version updates Section 6.2 to refer to draft-ietf-rtcweb-fec.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, November 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-21.txt).
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This version makes a number of changes to address comments made at the IETF 91 meeting in Honolulu:
- In Section 4.3, clarify reference to draft-ietf-rtcweb-audio and add reference draft-ietf-rtcweb-video for mandatory to implement codec and payload formats. Clarify that any other codec and payload format MAY be implemented, and that all payload formats MUST be signalled before use.
- In Section 5.2.5, note that the CVO header extension MUST be implemented as described in Section 4 of draft-ietf-rtcweb-video.
- In Section 4.1, note that the RTCP SDES MID item MUST be implemented if the SDP bundle extension is used. In Section 5.2.4, note that the RTP MID header extension MUST be implemented if the SDP bundle extension is used.
- In Sections 5.2.1, 5.2.2, and 5.2.3, clarify that the header extension needs to be negotiated before it can be used.
- In Section 5.2.2, clarify that the RFC 6464 client-to-mixer audio level header extension MUST be implemented.
- Update Section 6.2 on FEC to refer to draft-uberti-rtcweb-fec.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, November 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-20.txt).
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This version updates the discussion RTP retransmission to require receivers to implement support for RTP retransmission sent using SSRC multiplexing, while making support for session multiplexing optional.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, October 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-19.txt).
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This version clarifies that both session- and SSRC-multiplexing need to be supported for the RTP retransmission payload format.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, October 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-18.txt).
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This version attempts to align the terminology with the most recent version of the rtcweb-overview draft.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, August 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-17.txt).
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In Section 4.1 change "RTCP is a fundamental and integral part of RTP, and MUST be implemented in all WebRTC applications" to "...MUST be implemented and used...".
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, July 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-16.txt).
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This is a minor updated, based on discussion at IETF 90, that adds a reminder about discontinuous transmission to Section 4.1, and makes a clarification about use of multiple RTCP CNAMEs in Section 11.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, May 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-15.txt).
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This version updates the draft to reflect comments received at the joint IETF/W3C WebRTC interim meeting held in Washington DC in May 2014. The changes are as follows:
- make support for RTP multi-stream optimisations "MAY support, MUST signal before use";
- give a reference to draft-ietf-avtcore-rtp-multi-stream to justify the recommendation to set T_rr_interval = 4 when using the RTP/SAVPF profile;
- clarify that implementations MUST signal extensions before use (although they need to be robust to non-signalled extensions); and
- discuss mitigation of potential denial of service attacks due to malicious configuration of RTCP parameters.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, May 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-14.txt).
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This version is intended to address working group last call comments. The changes are summarised in a presentation on WebRTC: RTP Usage WG Last Call Comments that Magnus Westerlund will give at the IETF/W3C WebRTC interim meeting in May 2014.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, April 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-13.txt).
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Changes in this version of the draft include:
- clarify terminology;
- clarify that use of the reduced RTCP reporting interval described in Section 6.2 of RFC 3550 is RECOMMENDED;
- clarify that RTP payload type numbers MUST NOT be assigned to different RTP payload formats, or different configurations of the same RTP payload format, with a single RTP session (noting that an SDP bundle group forms a single RTP session);
- clarify choice of RTCP CNAME;
- clarify use of different RTP topologies in Section 5.1;
- clarify behaviour around encryption of RTP header extensions in sections 5.2.2, 5.2.3, and 13;
- clarify handling of leap seconds, by reference to RFC 7164; and
- provide an expanded example showing how RTCP feedback can be used for congestion control in Section 7.2.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, February 2014, Work in progress (draft-ietf-rtcweb-rtp-usage-12.txt).
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This version of the draft makes the following changes:
- the use of the SLI and RPSI feedback messages is made RECOMMENDED rather than OPTIONAL, based on list discussion;
- section 5.2.4 on Associating RTP Media Streams and Signalling Contexts has been removed, since this should be specified as part of the WebRTC signalling specification;
- defer to the WebRTC security architecture draft for details of the cipher suites, DTLS-SRTP protection profiles, key management, etc.;
- remove reference to the shim-based approach to running multiple RTP sessions on a single transport-layer flow;
- clarified the API and the MS and MST mappings to SSRC existence;
- put requirement on the CSRC list API in the W3C;
- removed most of the discussion of simulcast, leaving only some comments in Section 12.1 regarding simulcast with existing functionality; and
- slightly expand discussion of the implications of per-packet DiffServ marking on RTP streams.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, December 2013, Work in progress (draft-ietf-rtcweb-rtp-usage-11.txt).
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The -11 version of this draft addresses the outcomes of the discussions at IETF 88 in Vancouver. The changes are:
- Handling multiple synchronization contexts (CNAMES)
- WebRTC API mapping, based on the current working draft API from the W3C. There are proposals to remove the MediaStream concept from that API. If that occurs, this draft will have to be updated to match.
- Forwarding of MediaStreamTracks from one PeerConnection to another are now discussed, along with the need to be able to handle that as a MediaSource, i.e. be capable of transcoding the media to suit outgoing requirements, including bit-rate adaptation.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, October 2013, Work in progress (draft-ietf-rtcweb-rtp-usage-10.txt).
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The -10 version of thus draft was submitted just prior to the cut-off for discussion at IETF 88. The main changes in this version are as follows:
- Clarify that keying for RTP/SAVPF profile specified in the security-arch draft
- Clarify that an endpoint can have multiple RTCP CNAMEs if it sends streams synchronised to multiple clocks
- Clarify that the RTP circuit breaker is a boundary condition, and that applications also need to implement congestion control
- Clarify that RTP/AVPF + DTLS-SRTP keying is mandatory to implement
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, September 2013, Work in progress (draft-ietf-rtcweb-rtp-usage-09.txt).
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This version just updates the references, to catch-up with recently published RFCs.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, September 2013, Work in progress (draft-ietf-rtcweb-rtp-usage-08.txt).
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The changes in this version are as follows:
- Reorganise and update Section 12
- Remove most of the Appendix, with the exception of sections A.5 and A.6, which have been merged into Section 12.
- Expand the (tbd: ...) regarding draft-westerlund-mmusic-max-ssrc-01 in Section 4.1
- In Section 4.3, clarify that the mapping between RTP payload types and specific configurations of payload formats MUST be agreed before those payload types/formats can be used. SDP "a=rtpmap:" and "a=fmtp:" lines provide a way to do this.
- In section 4.4, clarify that participants must agree to use multiple types of media in a single RTP session, and that BUNDLE an example of how this can be done.
- In Section 5.1, remove bullet stating that Transport Translators are not of immediate interest; there's no real reason why they can't be used.
- Add a placeholder section 5.2.4, on Associating RTP Media Streams and Signalling Contexts.
- Update references
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, July 2013, Work in progress (draft-ietf-rtcweb-rtp-usage-07.txt).
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The changes in this version are as follows:
- In Section 4.1, expand discussion of multiple simultaneous SSRC values in a single RTP session, referencing draft-ietf-avtcore-rtp-multi-stream and draft-ietf-avtcore-rtp-multi-stream-optimisation. Add a note suggesting that draft-westerlund-mmusic-max-ssrc might be a useful addition.
- In Section 4.3, expand discussion of RTP payload type assignment, and briefly explain how the RTP payload type can be used to associate an RTP media stream with a signalling context (e.g., an SDP "m=" line).
- In Section 4.3, clarify that this memo does not specify mandatory to implement codecs or RTP payload formats. Also clarify that any codec where there is an RTP payload format and SDP offer/answer procedures defined can be used with WebRTC, provided it is negotiated.
- In Section 4.4, clarify that a single RTP session is to be used for all RTP media streams. Note that there is no consensus to use a shim-based approach.
- In Section 4.8, clarify that implementations MUST be prepared to accept RTP and RTCP packets using SSRCs that have not been explicitly signalled. Briefly explain how RTP media streams can be associated with a signalling context, using either a signalled SSRC or the RTP payload type.
- In Section 5.2.1, clarify that the rapid synchronisation extensions are in addition to RTCP SR-based synchronisation, and do not replace it.
- In Section 6.1, add a reference to RFC 3550 to explain how to do the RTT calculation from SR/RR packets.
- Remove the first paragraph of Section 7.2, since the issue is covered by the text in the last paragraph of Section 7.1.
- Move the remaining content from Section 7.2 to the end of Section 7.4, where it fits better.
- In what was Section 7.4, and is now Section 7.3, add some words about interoperability between sender- and receiver-driven congestion control. This issue is not solvable in this draft, since we don't yet have any standardised congestion control algorithms of either type, however it is important to note that it needs to be addressed when future congestion control algorithms are defined.
- Rewrite Section 8, on Performance Monitoring, to reflect the discussion at IETF 86.
- Add a note to Section 12.2 about the issue #20 in the RTCWEB issue tracker
- Editorial fixes to Sections 1, 2, 4.2, 4.5, 4.6, 4.7, 5.1, 6, 7, 10, and 11.
- Update list of open issues.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-ietf-rtcweb-rtp-usage-06.txt).
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The changes in this version of the draft are as follows:
- Expand and clarify discussion of RTP session multiplexing in Section 4.4
- Add Section 7.2 on RTCP extensions for congestion control
- Clarify Section 12.1 on RTP Sessions and PeerConnections
- Expand Section 12.4 on SSRC collision detections
- Rewrite and clarify Section 12.5 on Contributing Sources and the CSRC list
- Rewrite and clarify Section 12.9 on differentiated treatment of flows
- Expand security considerations
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, October 2012, Work in progress (draft-ietf-rtcweb-rtp-usage-05.txt).
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The changes in this version of the draft are as follows:
- Use RFC 2119 terminology by reference, rather than by copying the definitions.
- In Section 4.2 on the Choice of RTP Profile, note that the RTP/SAVPF profile with the updated list of recommended codecs is mandated, not the standard RTP/SAVPF profile. Update Section 4.3 on the Choice of RTP Payload Formats to match.
- In Section 4.6, clarify that the use of non-compound RTCP packets MUST be negotiated on the signalling channel before use, and that implementations are REQUIRED to support compound RTCP feedback packets if the remote endpoint does not agree to use non-compound RTCP packets.
- In Section 4.9, remove the reference to RFC 6222 and instead reference the RFC 622bis draft.
- Update references to RFC 5117 to joint to the RTP Topologies update draft.
- In Section 5.1.1, clarify that a WebRTC sender is REQUIRED to understand and react to FIR messages it receives, but that sending FIR messages is OPTIONAL.
- Rewrite Section 7 on rate control and media adaptation for clarity. Merge the previous Sections 7.1 and 7.2 into a single new section, and try to better explain the relationship between the RTP circuit breakers, the signalled SDP bandwidth limitations, and any RTP/AVPF TMMBR messages.
- Add Section 13 on Open Issues.
- Revise and expand Section 15 on Security Considerations.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, July 2012, Work in progress (draft-ietf-rtcweb-rtp-usage-04.txt).
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This version attempts to reflect the outcomes of the IETF RTCWeb working group interim meeting that was held in Kista on 12-13 June 2012. This is a major update to the draft, touching almost every section of the text. The changes are too numerous to describe in detail, but a complete diff from the previous version is available. After the extensive discussion in the interim meeting, we are not planning to present this draft in the IETF 84 meeting in Vancouver; discussion should take place on the IETF RTCWeb working group's mailing list instead. If there are still unresolved issues, we will discuss the draft further at IETF 85 in Atlanta in November 2012.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, June 2012, Work in progress (draft-ietf-rtcweb-rtp-usage-03.txt).
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This version of the draft is submitted for discussion at the IETF RTCWeb working group's interim meeting in Kista on 12-13 June 2012. This new version includes a greatly expanded discussion of RTP topologies, some more concrete recommendations on use of the core RTP protocol, and various other clarifications throughout. This is still work in progress, and will likely evolve significantly after the discussion at the interim meeting.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, March 2012, Work in progress (draft-ietf-rtcweb-rtp-usage-02.txt).
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We've just posted an update to our draft on the use of RTP in the RTCWeb context. This is a minor update, consisting almost entirely of editorial clean-ups. The only technical changes are to make RFC 6222 support REQUIRED rather than RECOMMENDED, and to replace the discussion of congestion control requirements with references to Randell Jesup's draft on Congestion Control Requirements For Real Time Media, and my new draft on RTP Congestion Control: Circuit Breakers for Unicast Sessions.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, October 2011, Work in progress (draft-ietf-rtcweb-rtp-usage-01.txt).
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We've updated our draft on RTP Requirements for RTC-Web, and renamed it to better reflect the content. Changes in this version include:
- Update title, rewrite Abstract and Introduction, and restructure the draft
- Update section on Rate Control and Media Adaptation
- Update section on Security Considerations
- Update section on Use of RTP: Core Protocols
- Add some initial discussion of multiplexing several RTP sessions onto a single lower-layer transport, primarily referencing other drafts for the content.
- Address comments by Harald Alvestrand from the mailing list (29 August 2011)
- Colin Perkins, Magnus Westerlund, and Jörg Ott, RTP Requirements for RTC-Web (.txt|.pdf), Internet Engineering Task Force, September 2011, Work in progress (draft-ietf-rtcweb-rtp-usage-00.txt).
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This draft was accepted as a working group draft after the RTCWeb interim meeting. This version updates the draft filename and date to reflect that, but makes no other changes.
This draft replaces draft-perkins-rtcweb-rtp-usage